[asterisk-users] ERROR during high volume MoH dialplan

[Joseph Smith]

Thanks for the suggestion Tony,


I installed each codec for MoH, core sounds, and extra sound packages.  Unfortunately the tests produce the same results.

[Sep  1 20:36:45] ERROR[10081][C-00007fe5]: frame.c:343 ast_frdup: FRACK!, Failed assertion Excessive refcount 100000 reached on ao2 object 0x20380b0 (

continuously for a while followed by a

[Sep  1 20:36:46] WARNING[7761][C-0000770d]: taskprocessor.c:888 taskprocessor_push: The 'subp:PJSIP/sipp-00000020' task processor queue reached 500 scheduled tasks.

Then this time Asterisk actually crashed. :(

________________________________
From: asterisk-users-bounces at lists.digium.com <asterisk-users-bounces at lists.digium.com> on behalf of Tony Mountifield <tony at softins.co.uk>
Sent: Friday, September 1, 2017 11:01 AM
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] ERROR during high volume MoH dialplan

In article <CY4PR2201MB14643C2177C953FA27AC9E2BA8920 at CY4PR2201MB1464.namprd22.prod.outlook.com>,
Joseph Smith <warlock1999 at hotmail.com> wrote:
>
> Thanks for the feedback.
>
> I do agree with having multiple smaller servers.  When I was first approached with this task I mentioned as much.
> However, the current desire is to work with already existing hardware.  That is out of my hands at the moment unless it
> just can't be done.  I will explore Freeswitch a bit soon to compare it as well.
>
>
> I am struggling to find what the bottle neck is in this scenario.  Does anyone have any advice on what that could be or
> on steps to discover it?   Do you think that tasks are pooling up because of transcoding?  If so would it help to change
> the codec that is being used?  I am not sure about the MoH but the audio files I am using are gsm.

You will find it less taxing on the server if you have MoH files and sounds files
available in all the possible native formats. Then Asterisk can use the appropriate
one for the channel without transcoding.

On my systems I have MoH and sounds installed in wav, ulaw, alaw, gsm and g729.

They will also sound better than transcoding from the gsm versions.

Cheers
Tony
--
Tony Mountifield
Work: tony at softins.co.uk - http://www.softins.co.uk
Software Insight - Welcome<http://www.softins.co.uk/>
www.softins.co.uk
Welcome. Software Insight Ltd is a small but expert company specialising in software and systems development and systems administration. We pride ourselves in ...



Play: tony at mountifield.org - http://tony.mountifield.org
[http://tony.mountifield.org/images/tony2.jpg]<http://tony.mountifield.org/>

Tony Mountifield's Home Page<http://tony.mountifield.org/>
tony.mountifield.org
Tony Mountifield's Home Page. This page is still under construction (despite having been started a long time ago!) It will grow as I think of more things to put in ...




--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
api digital - problem solved.<http://www.api-digital.com/>
www.api-digital.com
API Digital Website




Check out the new Asterisk community forum at: https://community.asterisk.org/
[https://community.asterisk.org/images/default-apple-touch-icon.png]<https://community.asterisk.org/>

Asterisk Community<https://community.asterisk.org/>
community.asterisk.org
The Asterisk Community's home for Discussion ... Is there any way to share same queue with same agents between multiple servers in a multiple server Asterisk ...




New to Asterisk? Start here:
      https://wiki.asterisk.org/wiki/display/AST/Getting+Started
Getting Started - Asterisk Project - Asterisk Project Wiki<https://wiki.asterisk.org/wiki/display/AST/Getting+Started>
wiki.asterisk.org
When learning Asterisk it is important to start off on the right foot, so this section of the wiki covers orientation for learning Asterisk as well as installation ...




asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20170901/f84b1ef8/attachment.html>